Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 370 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc
new file mode 100644
index 0000000000..5f2b4872b9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc
@@ -0,0 +1,370 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/gain_control_impl.h"
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/agc/legacy/gain_control.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+typedef void Handle;
+
+namespace {
+int16_t MapSetting(GainControl::Mode mode) {
+  switch (mode) {
+    case GainControl::kAdaptiveAnalog:
+      return kAgcModeAdaptiveAnalog;
+    case GainControl::kAdaptiveDigital:
+      return kAgcModeAdaptiveDigital;
+    case GainControl::kFixedDigital:
+      return kAgcModeFixedDigital;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return -1;
+}
+
+// Applies the sub-frame `gains` to all the bands in `out` and clamps the output
+// in the signed 16 bit range.
+void ApplyDigitalGain(const int32_t gains[11],
+                      size_t num_bands,
+                      float* const* out) {
+  constexpr float kScaling = 1.f / 65536.f;
+  constexpr int kNumSubSections = 16;
+  constexpr float kOneByNumSubSections = 1.f / kNumSubSections;
+
+  float gains_scaled[11];
+  for (int k = 0; k < 11; ++k) {
+    gains_scaled[k] = gains[k] * kScaling;
+  }
+
+  for (size_t b = 0; b < num_bands; ++b) {
+    float* out_band = out[b];
+    for (int k = 0, sample = 0; k < 10; ++k) {
+      const float delta =
+          (gains_scaled[k + 1] - gains_scaled[k]) * kOneByNumSubSections;
+      float gain = gains_scaled[k];
+      for (int n = 0; n < kNumSubSections; ++n, ++sample) {
+        RTC_DCHECK_EQ(k * kNumSubSections + n, sample);
+        out_band[sample] *= gain;
+        out_band[sample] =
+            std::min(32767.f, std::max(-32768.f, out_band[sample]));
+        gain += delta;
+      }
+    }
+  }
+}
+
+}  // namespace
+
+struct GainControlImpl::MonoAgcState {
+  MonoAgcState() {
+    state = WebRtcAgc_Create();
+    RTC_CHECK(state);
+  }
+
+  ~MonoAgcState() {
+    RTC_DCHECK(state);
+    WebRtcAgc_Free(state);
+  }
+
+  MonoAgcState(const MonoAgcState&) = delete;
+  MonoAgcState& operator=(const MonoAgcState&) = delete;
+  int32_t gains[11];
+  Handle* state;
+};
+
+int GainControlImpl::instance_counter_ = 0;
+
+GainControlImpl::GainControlImpl()
+    : data_dumper_(new ApmDataDumper(instance_counter_)),
+      mode_(kAdaptiveAnalog),
+      minimum_capture_level_(0),
+      maximum_capture_level_(255),
+      limiter_enabled_(true),
+      target_level_dbfs_(3),
+      compression_gain_db_(9),
+      analog_capture_level_(0),
+      was_analog_level_set_(false),
+      stream_is_saturated_(false) {}
+
+GainControlImpl::~GainControlImpl() = default;
+
+void GainControlImpl::ProcessRenderAudio(
+    rtc::ArrayView<const int16_t> packed_render_audio) {
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    WebRtcAgc_AddFarend(mono_agcs_[ch]->state, packed_render_audio.data(),
+                        packed_render_audio.size());
+  }
+}
+
+void GainControlImpl::PackRenderAudioBuffer(
+    const AudioBuffer& audio,
+    std::vector<int16_t>* packed_buffer) {
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
+  std::array<int16_t, AudioBuffer::kMaxSplitFrameLength>
+      mixed_16_kHz_render_data;
+  rtc::ArrayView<const int16_t> mixed_16_kHz_render(
+      mixed_16_kHz_render_data.data(), audio.num_frames_per_band());
+  if (audio.num_channels() == 1) {
+    FloatS16ToS16(audio.split_bands_const(0)[kBand0To8kHz],
+                  audio.num_frames_per_band(), mixed_16_kHz_render_data.data());
+  } else {
+    const int num_channels = static_cast<int>(audio.num_channels());
+    for (size_t i = 0; i < audio.num_frames_per_band(); ++i) {
+      int32_t sum = 0;
+      for (int ch = 0; ch < num_channels; ++ch) {
+        sum += FloatS16ToS16(audio.split_channels_const(kBand0To8kHz)[ch][i]);
+      }
+      mixed_16_kHz_render_data[i] = sum / num_channels;
+    }
+  }
+
+  packed_buffer->clear();
+  packed_buffer->insert(
+      packed_buffer->end(), mixed_16_kHz_render.data(),
+      (mixed_16_kHz_render.data() + audio.num_frames_per_band()));
+}
+
+int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) {
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
+  RTC_DCHECK_EQ(audio.num_channels(), *num_proc_channels_);
+  RTC_DCHECK_LE(*num_proc_channels_, mono_agcs_.size());
+
+  int16_t split_band_data[AudioBuffer::kMaxNumBands]
+                         [AudioBuffer::kMaxSplitFrameLength];
+  int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
+      split_band_data[0], split_band_data[1], split_band_data[2]};
+
+  if (mode_ == kAdaptiveAnalog) {
+    for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+      capture_levels_[ch] = analog_capture_level_;
+
+      audio.ExportSplitChannelData(ch, split_bands);
+
+      int err =
+          WebRtcAgc_AddMic(mono_agcs_[ch]->state, split_bands,
+                           audio.num_bands(), audio.num_frames_per_band());
+
+      if (err != AudioProcessing::kNoError) {
+        return AudioProcessing::kUnspecifiedError;
+      }
+    }
+  } else if (mode_ == kAdaptiveDigital) {
+    for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+      int32_t capture_level_out = 0;
+
+      audio.ExportSplitChannelData(ch, split_bands);
+
+      int err =
+          WebRtcAgc_VirtualMic(mono_agcs_[ch]->state, split_bands,
+                               audio.num_bands(), audio.num_frames_per_band(),
+                               analog_capture_level_, &capture_level_out);
+
+      capture_levels_[ch] = capture_level_out;
+
+      if (err != AudioProcessing::kNoError) {
+        return AudioProcessing::kUnspecifiedError;
+      }
+    }
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
+                                         bool stream_has_echo) {
+  if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
+    return AudioProcessing::kStreamParameterNotSetError;
+  }
+
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+                audio->num_frames_per_band());
+  RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
+
+  stream_is_saturated_ = false;
+  bool error_reported = false;
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    int16_t split_band_data[AudioBuffer::kMaxNumBands]
+                           [AudioBuffer::kMaxSplitFrameLength];
+    int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
+        split_band_data[0], split_band_data[1], split_band_data[2]};
+    audio->ExportSplitChannelData(ch, split_bands);
+
+    // The call to stream_has_echo() is ok from a deadlock perspective
+    // as the capture lock is allready held.
+    int32_t new_capture_level = 0;
+    uint8_t saturation_warning = 0;
+    int err_analyze = WebRtcAgc_Analyze(
+        mono_agcs_[ch]->state, split_bands, audio->num_bands(),
+        audio->num_frames_per_band(), capture_levels_[ch], &new_capture_level,
+        stream_has_echo, &saturation_warning, mono_agcs_[ch]->gains);
+    capture_levels_[ch] = new_capture_level;
+
+    error_reported = error_reported || err_analyze != AudioProcessing::kNoError;
+
+    stream_is_saturated_ = stream_is_saturated_ || saturation_warning == 1;
+  }
+
+  // Choose the minimun gain for application
+  size_t index_to_apply = 0;
+  for (size_t ch = 1; ch < mono_agcs_.size(); ++ch) {
+    if (mono_agcs_[index_to_apply]->gains[10] < mono_agcs_[ch]->gains[10]) {
+      index_to_apply = ch;
+    }
+  }
+
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    ApplyDigitalGain(mono_agcs_[index_to_apply]->gains, audio->num_bands(),
+                     audio->split_bands(ch));
+  }
+
+  RTC_DCHECK_LT(0ul, *num_proc_channels_);
+  if (mode_ == kAdaptiveAnalog) {
+    // Take the analog level to be the minimum accross all channels.
+    analog_capture_level_ = capture_levels_[0];
+    for (size_t ch = 1; ch < mono_agcs_.size(); ++ch) {
+      analog_capture_level_ =
+          std::min(analog_capture_level_, capture_levels_[ch]);
+    }
+  }
+
+  if (error_reported) {
+    return AudioProcessing::kUnspecifiedError;
+  }
+
+  was_analog_level_set_ = false;
+
+  return AudioProcessing::kNoError;
+}
+
+// TODO(ajm): ensure this is called under kAdaptiveAnalog.
+int GainControlImpl::set_stream_analog_level(int level) {
+  data_dumper_->DumpRaw("gain_control_set_stream_analog_level", 1, &level);
+
+  was_analog_level_set_ = true;
+  if (level < minimum_capture_level_ || level > maximum_capture_level_) {
+    return AudioProcessing::kBadParameterError;
+  }
+  analog_capture_level_ = level;
+
+  return AudioProcessing::kNoError;
+}
+
+int GainControlImpl::stream_analog_level() const {
+  data_dumper_->DumpRaw("gain_control_stream_analog_level", 1,
+                        &analog_capture_level_);
+  return analog_capture_level_;
+}
+
+int GainControlImpl::set_mode(Mode mode) {
+  if (MapSetting(mode) == -1) {
+    return AudioProcessing::kBadParameterError;
+  }
+
+  mode_ = mode;
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK(sample_rate_hz_);
+  Initialize(*num_proc_channels_, *sample_rate_hz_);
+  return AudioProcessing::kNoError;
+}
+
+int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
+  if (minimum < 0 || maximum > 65535 || maximum < minimum) {
+    return AudioProcessing::kBadParameterError;
+  }
+
+  minimum_capture_level_ = minimum;
+  maximum_capture_level_ = maximum;
+
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK(sample_rate_hz_);
+  Initialize(*num_proc_channels_, *sample_rate_hz_);
+  return AudioProcessing::kNoError;
+}
+
+int GainControlImpl::set_target_level_dbfs(int level) {
+  if (level > 31 || level < 0) {
+    return AudioProcessing::kBadParameterError;
+  }
+  target_level_dbfs_ = level;
+  return Configure();
+}
+
+int GainControlImpl::set_compression_gain_db(int gain) {
+  if (gain < 0 || gain > 90) {
+    RTC_LOG(LS_ERROR) << "set_compression_gain_db(" << gain << ") failed.";
+    return AudioProcessing::kBadParameterError;
+  }
+  compression_gain_db_ = gain;
+  return Configure();
+}
+
+int GainControlImpl::enable_limiter(bool enable) {
+  limiter_enabled_ = enable;
+  return Configure();
+}
+
+void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) {
+  data_dumper_->InitiateNewSetOfRecordings();
+
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
+             sample_rate_hz == 48000);
+
+  num_proc_channels_ = num_proc_channels;
+  sample_rate_hz_ = sample_rate_hz;
+
+  mono_agcs_.resize(*num_proc_channels_);
+  capture_levels_.resize(*num_proc_channels_);
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    if (!mono_agcs_[ch]) {
+      mono_agcs_[ch].reset(new MonoAgcState());
+    }
+
+    int error = WebRtcAgc_Init(mono_agcs_[ch]->state, minimum_capture_level_,
+                               maximum_capture_level_, MapSetting(mode_),
+                               *sample_rate_hz_);
+    RTC_DCHECK_EQ(error, 0);
+    capture_levels_[ch] = analog_capture_level_;
+  }
+
+  Configure();
+}
+
+int GainControlImpl::Configure() {
+  WebRtcAgcConfig config;
+  // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
+  //            change the interface.
+  // RTC_DCHECK_LE(target_level_dbfs_, 0);
+  // config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
+  config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
+  config.compressionGaindB = static_cast<int16_t>(compression_gain_db_);
+  config.limiterEnable = limiter_enabled_;
+
+  int error = AudioProcessing::kNoError;
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    int error_ch = WebRtcAgc_set_config(mono_agcs_[ch]->state, config);
+    if (error_ch != AudioProcessing::kNoError) {
+      error = error_ch;
+    }
+  }
+  return error;
+}
+}  // namespace webrtc